Asterisk flash based sip phonepekerjaan

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    16,510 asterisk flash based sip phone tugasan ditemui, harga dalam USD

    Saya membutuhkan yg paham dan pengalaman utk melakukan setting/configurasi asterisk saat ini sdh terpasang akan tetapi masih mengkonsumsi cpu server boros sekali rencanaakan dipakai utk call center dg jumlah agent 100 orang.

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    Country Sales Manager UK 6 hari left
    DISAHKAN

    ...of the year, we are going to launch a SaaS in the UK. Target group are delivery restaurants without website that allows ordering. We provide leads for a start, dedicated SIP phone numbers, a CRM, and training. We offer high sales commissions and contract extension commissions. We expect impeccable UK accent - no exceptions. We are looking for a Country

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    ...of the year, we are going to launch a SaaS in the UK. Target group are delivery restaurants without website that allows ordering. We provide leads for a start, dedicated SIP phone numbers, a CRM, and training. We offer high sales commissions and contract extension commissions. We expect impeccable UK accent - no exceptions. When you apply make sure

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    6 bida

    ...Click to Call (CTI) - Click in contact in PipeDrive and call to customer. Call History - All calls made will be logged in the customer details of Pipedrive. Have to match the phone number from Freepbx into Pipedrive. Regardless if the number is in format +1-xxx-xxx-xxxx or xxx-xxx-xxxx Call recording - All calls are currently recorded on Freepbx. Need

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    asterisk job 6 hari left
    DISAHKAN

    asterisk person to work on an existing product to fix and update the product knowledge needed: PHP Asterisk database

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    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

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    ...his SIP server and we need to create separate account for each gateway . and calls should send to specific termination . call should pass with sip , g729 or g723. in client end we want to setup asterisk module which can convert calls from sip to sip and pass the calls to gateway . 01. asterisk or SBO server. which receive calls from many sip se...

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    I have an Asterisk server for private use already running, and would like to acidify a Trunk using an FXO VoIP Gateway, for this is necessary to create a sip trunk in Asterisk and I do not know how to do. In my attempts, I can even connect SIP between them, but I can not complete calls.

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    Preciso que configure um ATCOM Modelo IP2G4A como Trunk de um PBX Asterisk Issabel que já está instalado. Todos estão na mesma LAN já com a rede configurada.

    $10 - $30
    $10 - $30
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    I need a server administrator with experience on VoIP technology. FreeBPX, Asterisk, SIP Clients Cisco SIP phone provisioning and some other SIP phones. Developer most commute to office in Rupnaghar India. If you don't leave in India please don't apply.

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    Tempatan
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    Freepbx and Asterisk settings 5 hari left
    DISAHKAN

    I am using for our office a pbx with regular sip phones and some softphones. The softphones work using Bria Mobile with Push notifications enabled. However, the phone doesn’t work properly on background meaning that push notifications don’t work as expected for receiving calls. Here is the page where Bria mobile explains the settings that needs to be

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    Hola con todos Alguien ha podido integar infusionsoft con centrales telefonicas ip como Elastix or Supermicro-Asterisk. La integración debe permitir : - Conocer duración de llamada - Grabar llamada - Origen de la llamada - Numero de llamadas por Gestor y por campaña de llamadas - Consolidación general de las llamadas Para controlar al gestor de

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    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

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    ...nr1: comes through normal ISDN to IP PBX Set nr2: comes through incoming Vitual Voip DID. All the numbers of set nr 2 work but i have 1 nr that does not work as i get error sip 503. For security reasons , you will be operating through Teamviewer on my computer which is connected to the remote site of my client where the IP PBX is. I also need to check

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    ...Tabelle (ca. 20 Wörter) Total geht es um ca. 100 Texte in diesem Format zu verschiedenen Hunderassen. Preis also für 100 Texte angeben. Bitte starte dein Angebot mit einem Asterisk (*), um zu zeigen, dass du das Inserat gelesen hast....

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    Looking for someone to pass asterisk logs to, to determine why some calls are dropped or why calls are not routing properly.

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    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

    $836 (Avg Bid)
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    10 bida
    VOIP yate call pass 3 hari left
    DISAHKAN

    i need to install yate on openwrt and pass calls server to my gateway we pass call useing sip to sip if you can make it please bid

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    app development 3 hari left

    Android sip cellular gateway we are looking for an expert in developing mobile app to develop an app that will expect voice calls using usip server and dial out local number using the mobile network

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    integrate bitrix with asterisk system -

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    ...modify those. Below are more detailed descriptions of the api/ and frontend/ directories. Locations that you will be mainly focussing on as you develop are marked with an asterisk (*). Backend API (api/) api ├── Dockerfile # Description of Docker image ├── [login to view URL] # NPM dependencies ├── spec # The backend tests

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    Hi, I need a wordpress plugin which will allow the users to place a call on the website. I will be integrating a sip gateway for same. Users can place free calls with some restrictions like 1 minutes, or a 10 seconds ad after every 1 minute. Paying users can place unlimited calls until their credits are exhausted I wish to achieve a website like https://ievaphone

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    9 bida

    Need someone to configure Asterisk secure websocket for javascript library integration (such as jssip or SIPml). Asterisk is installed. Access will be provided.

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    Actionscript 3.0 SIP 2 hari left
    DISAHKAN

    1. REGISTER + SUBSCRIBE to SIP server, with authentication 2. Accept INVITES 3. play inbound SIP packets and convert microphone to outbound sip That's it. If you've done it before, it's 1 hour work. Fixed payment $100. We need Actionscript in AIR.

    $30 - $250
    $30 - $250
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    PHP SIP client 2 hari left
    DISAHKAN

    PHP SIP client: 1. REGISTER + SUBSCRIBE to SIP server, WITH authentication 2. Accept INVITEs to send only, NO recv (so no traffic inbound) 3. Return WAV file to caller; depending on called number either file 1 or file 2 4. BYE to disconnect, that's it. - FIXED $40 for working code (do NOT ASK FOR MORE) - It's not even 30 minutes work if you've done

    PHP
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    VOIP Project 2 hari left

    Signalling work with CISCO CUCM Understanding VOIP - SIP (including Blind and Attended Transfer implementation) - VOIP Call analysis – Wire Shark or similar - Visual Basic Script language - Proprietary Cisco SIP protocol extensions - Cisco CUCM - Call flows of the Attended Call Transfer - Cisco Finesse handling of the Call Transfer Budget as outlined

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    Requirement two-way audio/video SIP based communications form the mobile phones to Asterisks on ARM based targets and web browser targets

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    Tempatan
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    FreePBX support 2 hari left

    Hello freelancer, I currently need FreePBX installed with our asterisk instance. we have a 5 hardphones, 15 softphone users. We are in need of IVR setup, conferencing, outbound/inbound routes and trunks. Our freepbx will need to be setup based on best security practice and monitored for atleast 15 business days after installation. we are moving from

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    asterisk expert needed 2 hari left
    DISAHKAN

    Hi, i need assistance of an Asterisk expert on daily basis. I am looking for an experienced expert coder only. 5 star review will be provided for some one Good job! Happy Bidding!

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    Create SNOM Phone Dial Plan 2 hari left
    DISAHKAN

    I am looking for a SNOM phone guru that can create a dial plan for me with the following: 4 Digit Internal extension calling (extensions range from 2000-8000 & 9800) 7 Digit Local Calling 10 digit local calling (we are in Canada) 11 digit long distance calling Toll Free 800/855/844/866/877/888 International Calling with 011 Emergency calling with 911

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    i have voip device and i have small router it have 32mb ram os Openwrt i want to run 32calls so you need to use SIP2SIP register USA SIP server >>Bangladesh Openwrt Router>>>VOIP Device we cant use asterisk cz its need too much space i think best libre [login to view URL] or yate or besip [login to view URL]

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    add sip trunk to elastix i have sip trunk from STC

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    ...A: On Cloud And SIP Trunk has been created SSL is configured by "Let's Encrypt" Server B: On Local And SIP Trunk has been configured SSL is configured by "Self Signed" Result: Server A - SIP Trunk is appeared as not registered Server B - SIP Trunk is appeared as registered! I need to get all Registered, and the call is go through SIP...

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    configuracion de internos y de central asterisk

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    Hi, we need a professional Asterisk / freepbx sysadmin who can fix the nating in dmz using opnsense. currently we have done 90% of work Lan 192.168.X.0/24 WAN A.B.C.D (static public) DMZ 192.168.Y.0/24 PBX is here (also VIP to L.M.N.P public IP) using 1:1 Nating now the odd thing is that we are using linphone as softphone and we are having problem

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    ...documented installation and setup guide will also be required. I need you to develop some software for me. I would like this software to be developed for Linux using PHP, with Asterisk voip for call center. Solutions - Features - Call Features • Call Detail Records • Call Forward • Call Monitoring • Call Parking • Call Queuing • Call Recording • Call T...

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    To permit clients to fill out forms and documents on our...for the Nginx/Apache Web Server and the SMTP email Server, one for the MySQL Data Base Server and one for the module with LibreOffice. Invoice information comes from another Asterisk Server. To start, Teamviewer must be used, as some software is already [login to view URL] is a Linux only project !

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    we wish to have someone connect and configure our [login to view URL] to our SIP and Trunk

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    **Must speak both english and spanish** We are developing an inhouse an Asterisk based PBX solution to suit our needs. This is an ongoing project and we will need developers who are experts in PHP, NodeJS, CSS, MySQL, Optimization, Security. Bonus points for mobile development. We also need project managers. Previous experience managing development

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    I have a FreePBX/Asterisk System working at Amazon. I can access it directly or via a VPN. Normal telephony works as expected. 1) Trying to get WebRTC phone (via the UCP) working 2) Trying to integrate external WebRTC Can you help debug?

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    ...capability (CHECK on this) if not we will have to go open source with asterisk pbx, -Company with PBX need to be able to manage DID (phone numbers) and assign number to agents, administrator and super admin account, -Lower the cost per minute and per text (that is why we need to migrate to asterisk open source) -Lastly I need to be able to receive text message

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    SIP面板中找到100家从事房屋建筑的公司 如果事情不明确,请写信给我,我会尽力解释。 在中文里,我通过谷歌翻译写作。 我的母语是俄语。

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    We have an asterisk PBX integrated with Zoho CRM but it's delivering the channel ID instead of the dialed number to the the CRM extension. We need some one to fix the coding of the asterisk PBX to deliver the correct needed information

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    ...11-HSC branding removed 12-Phone & email support 13-Email marketing 14-Lists 15-Mobile optimization 16-Blog & content creation tools 17-SEO & content strategy 18-Social media 19-Calls-to-action 20-Landing pages 21-Analytics dashboards 22-Standard SSL certificate 23-Subdomain availability 24-Marketing automation 25-Goal-based nurturing 26-Salesforce integration

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    具备sip协议,支持30路以上fxs,可以实现点对点通信,需要提供相关技术支持

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    We Need somene that have some skills over ISSABEL Platform VOIP , Asterisk and Call Center Platform . We need to implemented some news features into reports database .

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    I want to create a ARI program that will play a message from URL in cloud and record the sentence spoken in channel and repeat what the person spoke after record is done Key thing here is I do not want any button pressing...Using ARI functions code should detetc when a person has spoken and when there is silence of more than 2 secs , consider the recording done Ideally i would like to implement t...

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    I need Linphone rebranded with our Company info, logo, colors and only allow the setup of SIP accounts over TLS. Removal of options to create a linphone account. This must be done on: IOS, Android, Windows and OSX The result must be packages ready to deploy through app stores. and source code must be handed over to us, as Linphone is open source

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    ...system must connect from carrier to carrier to allow VM to be placed on a number without ringing the phone. App will need to have standard features such as dashboards to see send data, user data, cost from carrier data. Must be able to connect to any SIP provider. Ringless Voicemail examples [login to view URL] [login to view URL]

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    Hi Dear Sir How Are You. Dear Sir I want to develop a soft switch from Asterisk PBX. Please Contact me and Give me your contact number Thanks [Removed by Freelancer.com Admin - please see Section 13 of our Terms and Conditions]

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